Selective response unit

ABSTRACT

In a telephone network, an incoming call is once received using an automatic voice response. A ring tone is not played immediately after the incoming. In the automatic response, an authorization sound that is necessary every time connection to the caller is made is generated each time by superimposing a background sound on a random number generated through random number generation processing. The caller is notified of the authorization sound. A ring tone is played in response to only incoming calls that return a valid response value, and not played in response to incoming calls that return an invalid response value.

CLAIM OF PRIORITY

The present application claims priority from Japanese patent applicationserial no. 2009-270292, filed on Nov. 27, 2009, the content of which ishereby incorporated by reference into this application.

TECHNICAL FIELD

The present invention relates to a selective response unit andparticularly to a selective response unit for blocking incoming callrequests from unwanted calls.

BACKGROUND OF THE INVENTION

With the spread of the Internet, IP telephone services using theInternet are being provided by telecommunications carriers and theInternet Service Providers (ISPs). Network forms in which analog andISDN telephone lines are connected to IP networks by use of MediaGateways (MGs) are also spreading. IP telephone services provideVoice-over-IP (VoIP) communications in which voice data is transferredin Real-time Transport Protocol (RTP) packets by establishing channels(sessions) between terminals before communications start.

As a session control protocol for establishing and disconnectingsessions between terminals, Session Initiation Protocol (SIP) isdefined, in which a session is controlled between terminals through anSIP server.

An SIP message includes a start line field and a header field forcarrying a request or response information and a message body field fordescribing contents of a session. For example, a Session DescriptionProtocol (SDP) is applied to describe contents of a session of an SIPmessage. In an SIP message, a communication party is identified using arequest Uniform Resource Identifier (URI) described in the start linefield. Additionally, in the SIP message, RTP packet communicationconditions between terminals such as protocols of, e.g., a call for acallee, voice, and video and a bit rate are defined in the process ofestablishment of a session.

With the spread of the Internet, unwanted e-mails transmittablesubstantially without cost are increasing. This is problem in society. Alarge number of unwanted e-mails are transmitted to mail addressescollected from bulletin boards and websites on the Internet andanticipated mail addresses. This is a burden on mail servers. Forexample, malicious unwanted e-mails use fake source addresses.

Like these unwanted e-mails, there is a concern that a large number ofunwanted calls using fake sources may be transmitted to collected andanticipated telephone numbers. In the past, unwanted calls wereperformed by persons. However, SPAM over IP Telephony (SPIT) usingInternet Protocols (IPs) is generated by automatically running SPAMgeneration software by use of a computer. Basically, in IP telephoneservices, when a caller requests connection to a callee, a ring tone isplayed. Therefore, there is a possibility that SPIT may be constantlyreceived as unwanted calls. Even when unwanted e-mails are received, thee-mails are first stored in a server and can be deleted without readingthem. However, since a ring tone is played in response to each receptionof an unwanted e-mail, there is a possibility that a callee may sufferfrom the ring tone 24 hours a day, throughout the year. Therefore, theproblem of unwanted calls is more serious than that of unwanted e-mails.

Communications with IP phones are also possible via MGs over analog andISDN telephone lines. Therefore, the same concern arises.

There are examples in which an automatic voice response is provided toprompt key inputs, e.g., for selecting services in an answer machine, afacsimile phone, and a call center. However, to provide services inthese examples, predetermined keys are announced by the automatic voiceresponse and inputs of the keys are received. Therefore, there is apossibility that the pattern may be easily learned and unwanted calls ofSPIT may be received.

In JP-A No. 2007-006467, a Turing test is performed to determine whethera caller transmits an unwanted call of SPIT by providing an SPIT filterbetween a caller and callee. In a Turing test using the SPIT filter, thedetermination is performed by comparing timing of response to questionsby an automatic response with an expected value of answers to thequestions.

In JP-A No. 2007-006467, however, the response timing is determinedusing a threshold and signal energy of the response. Additionally, noclear answer to be compared is shown in the determination using anexpected value of the answers. Accordingly, when ambiguous answers causefalse positives, SPIT bypasses the test and is then received.Additionally, multiple answer samples for automatic determination,sampling of signal energy, etc. need a high processing ability and alarge amount of memory. Further, an ambiguous answer causes a multistagetest, which bothers a caller.

SUMMARY OF THE INVENTION

In the present invention, in light of the above points, when there is anincoming call, a caller is notified of a random number sound, which isdifficult for machines to determine, and an automatic voice response istransmitted to the caller to prompt key inputs. Then, simple SPITdetermination is achieved by comparing a random number with a responsevalue.

According to one embodiment of the present invention, there is provideda selective response unit including a memory portion recording thereon acall control portion, a line interface portion, and an automatic voiceresponse table, and a voice processing circuit. The call control portionincludes an incoming-call authorization determination portion, aconnection-disconnection determination portion, and a random numbergeneration portion. When receiving a call connection request, theincoming-call authorization determination portion obtains a first voicecorresponding to a random number generated in the random numbergeneration portion from the automatic voice response table, andtransmits the first voice from the voice processing circuit to a sourceof the call connection request. When the voice processing circuit doesnot receive a second voice corresponding to a random number within apredetermined time, the connection-disconnection determination portiontransmits a call disconnection message to the source.

In the present invention, an incoming call is once received using anautomatic voice response without playing a ring tone immediately afterreception of the incoming call, and then a caller is notified of keysnecessary for connection. The keys correspond to authorization sound bygenerating a random number each time, changing a different samplingvoice for each digit, and overlapping a background noise with thevoices. This makes it difficult for callers of SPIT to learn of theresponse. Only when a key response is valid, a ring tone is played;otherwise, no ring tone is played. Therefore, an incoming call in whichannouncement starts unilaterally is disconnected before playing a ringtone.

Impersonations of callers are often made through personal servers.Therefore, a method is also effective in which an SIP server used by auser is set in advance and incoming calls from other than the SIP serverare rejected.

Further, measures to block unwanted calls by use of a whitelist ofnumbers of frequent callers and reliable callers and a blacklist ofnumbers of unwanted callers are increasing. However, in the measure toreject incoming calls of numbers of the black list, the number of URIsis infinite in IP phone and thus registration of URIs is limited. In themeasure to authorize only connection with incoming calls of numbers ofthe whitelist, incoming calls of public institutions and publictelephones are also rejected when their numbers are not registered inthe whitelist.

A user can also set, e.g., the following filtering instead of receivingall incoming calls by the automatic voice response:

(1) An automatic voice response to incoming calls from numbers of awhitelist is omitted.

(2) Incoming calls from numbers of a blacklist are rejected.

(3) A response is changed based on, e.g., whether an incoming call istransmitted from other than a registered server and a notification of anumber is transmitted.

The filtering is also applicable even when using Media Gateway ControlProtocol (MGCP), H.248, etc. instead of SIP as a session controlprotocol as well as when using SIP. In the following embodiments, an IPphone terminal is explained as a selective response unit, but may bemounted, e.g., in a media gateway or a server.

In response to an incoming call, SPIT can be easily filtered out bynotifying a caller of a random number to prompt key inputs by thecaller.

BRIEF DESCRIPTION OF THE DRAWINGS

Preferred embodiments of the present invention will now be described inconjunction with the accompanying drawings, in which;

FIG. 1 is a hardware block diagram of a structure of an IP phonenetwork;

FIG. 2 is a functional block diagram of a main portion of an IP phoneterminal;

FIG. 3 is a functional block diagram of a main portion of an SIP server;

FIG. 4A explains a session management table (No. 1);

FIG. 4B explains the session management table (No. 2);

FIG. 4C explains the session management table (No. 3);

FIG. 4D explains the session management table (No. 4);

FIG. 4E explains the session management table (No. 5);

FIG. 4F explains the session management table (No. 6);

FIG. 4G explains the session management table (No. 7);

FIG. 5 is a sequence of registering IP phone terminal information in anSIP server;

FIG. 6 is a sequence of connection, communication, and disconnectionbetween IP phone terminals;

FIG. 7A explains an INVITE message from an SIP server;

FIG. 7B explains the INVITE message from the SIP server;

FIG. 8 explains an automatic voice response table;

FIG. 9 is a flowchart of INVITE reception in an IP phone terminal;

FIG. 10 is a flowchart of ACK reception in an IP phone terminal;

FIG. 11A is a flowchart of random number generation in an IP phoneterminal;

FIG. 11B explains determination of the number of digits of a randomnumber in the random number generation;

FIG. 11C explains determination of a random number in the random numbergeneration;

FIG. 11D explains selection of sampling voices in the random numbergeneration;

FIG. 11E explains selection of background sounds in the random numbergeneration;

FIG. 11F explains a notification of an automatic voice response in therandom number generation;

FIG. 12 is a flowchart of wait for a response in an IP phone terminal;

FIG. 13 is a flowchart of determination in an IP phone terminal;

FIG. 14 is a sequence from incoming to disconnection of SPIT to and froman IP phone terminal;

FIG. 15 is a functional block diagram of a main portion of an IP phoneterminal having a filtering setting table;

FIG. 16A explains a filtering setting in an IP phone terminal to selectan operation for an incoming call from other than a registered SIPserver;

FIG. 16B explains the filtering setting in the IP phone terminal toselect an operation for an incoming call of a number not identified;

FIG. 16C explains the filtering setting in the IP phone terminal toselect an operation for an incoming call of an identified number;

FIG. 17 explains a structure of a registered-user management table;

FIG. 18 explains an SIP frame format;

FIG. 19 is a sequence when an incoming call is received from a numberregistered in a blacklist;

FIG. 20 is a sequence when an incoming call is received from a numberregistered in a whitelist;

FIG. 21 is a sequence in which the random number generation isperformed;

FIG. 22 is a flowchart of server determination;

FIG. 23 is a flowchart of the INVITE reception;

FIG. 24 is a flowchart of the ACK reception;

FIG. 25A explains a session management table when filtering is set (No.1);

FIG. 25B explains the session management table when the filtering is set(No. 2); and

FIG. 25C explains the session management table when the filtering is set(No. 3).

DESCRIPTION OF PREFERRED EMBODIMENTS

Hereafter, embodiments of the present invention are explained in detailin reference to the appended figures. Similar reference numerals denotegenerally similar components, and their explanation is not repeated.

Embodiment 1

First, with reference to FIG. 1, an IP phone network 100 using a SessionInitiation Protocol (SIP) as a session control protocol is explained. InFIG. 1, the IP phone network 100 includes multiple IP phone terminals 1,an SIP server (illustrated as SIP) 2, Layer-2-Switches (L2SWs) 3, threerouters (illustrated as R) 4, and a Media Gateway (MG) 5. The L2SW 3-1connects n IP phone terminals 1-1 n (n=A to N) to the router 4-1. TheL2SW 3-2 connects m IP phone terminals 1-2 m (m=A to M) to the router4-2. The three routers 4 form a ring type IP network 50. The router 4-1is connected to the MG 5. The router 4-2 is connected to the IP phoneterminal 1-3A. The router 4-3 is connected to the SIP server 2. The MG 5is connected to the IP phone terminal 1-4A.

The L2SWs 3 connect the IP phone terminals 1 to the routers 4. The MG 5connects a telephone switching network with the IP network 50.

In communications between the IP phone terminals 1, the caller IP phoneterminal 1 first registers with the SIP server 2. Next, the IP phoneterminal 1 controls connection by use of an SIP message through the SIPserver 2 to communicate with a communication party. Communications areestablished between the IP phone terminals 1 after establishment of asession connection.

With reference to FIG. 2, the main portion of the IP phone terminal 1 isexplained. In FIG. 2, the IP phone terminal 1 includes a line interface(IF) 11, a processor 12, a call control portion 13, a protocolprocessing portion 14, a session management table 18, an automaticresponse voice table 19, and user interfaces including a speaker 15, amicrophone 16, keys 17, and a voice processing circuit 10. The callcontrol portion 13 includes a registration portion 131, an incoming-callauthorization determination portion 133, a connection-disconnectionportion 132, and a random number generation portion 134.

The connection-disconnection portion 132 performs connection anddisconnection based on the incoming-call authorization determinationportion 133 and information inputted using the keys 17. Theconnection-disconnection portion 132 records a status of a call on thesession management table 18. When detecting an on-hook, theconnection-disconnection portion 132 records a disconnection time on thesession management table, and changes the status to disconnection. Theprotocol processing portion 14 decodes RTP packets received from theline IF 11. A decoded voice is outputted from the speaker 15. Theprotocol processing portion 14 encodes a voice inputted from themicrophone 16. The line IF 11 transmits the encoded RTP packets. Therandom number generation portion 134 generates random numbers, anddetermines sampling voices and background sounds. A voice processingcircuit 10 notifies a caller of a determined sampling voice andbackground sound. When detecting a sound of Push Button (PB), the speechprocessing circuit 10 converts the sound into digital data of 0-9, #,and *, and transmits them to the call control portion 13.

The registration portion 131 performs registration in the SIP server 2.The incoming-call authorization determination portion 133 determinesauthorization to an incoming call. The random number generation portion134 generates random numbers.

The main portion of the SIP server 2 is explained with reference to FIG.3. In FIG. 3, the SIP server 2 includes a line IF 21, a processor 22,and an SIP processing portion 23. The SIP processing portion 23 includesa REGISTER processing portion 231, an INVITE processing portion 232, aconnection-registration portion 233, and a disconnection-registrationportion 234. The REGISTER processing portion 231 performs REGISTERprocessing. The INVITE processing portion 232 performs INVITEprocessing. The connection-registration portion 233 performs connectionand registration processing.

The session management table 18 is explained with reference to FIGS. 4Ato 4G. In FIGS. 4A to 4G, the session management table 18 includes anincoming time 181, a disconnection time 182, a status 183, callerinformation 184, a comment 185, and a random number 186. The sessionmanagement table 18 is updated in INVITE processing,connection-registration processing, and disconnection-registrationprocessing.

FIGS. 4A to 4G are explained in detail later.

With reference to FIG. 5, REGISTER of session control through the SIPserver 2 to communicate between an IP phone terminal 1-1A and an IPphone terminal 1-2B is explained. In FIG. 5, for example when turned on,the IP phone terminal 1-2B performs registration to enablecommunications (S131). The IP phone terminal 1-2B transmits a REGISTERmessage to the SIP server 2 (S132). The SIP server 2 performs REGISTERprocessing (S133). The SIP server 2 transmits 200 OK to the IP phoneterminal 1-2B (S134).

Next, with reference to FIG. 6, processing for an incoming call requestbetween the IP phone terminal 1-1A and the IP phone terminal 1-2Bthrough the SIP server 2 is explained. In FIG. 6, the IP phone terminal1-1A transmits an INVITE message to the SIP server 2 (S101). The SIPserver 2 transmits the INVITE message to the IP phone terminal 1-2B(S102). The IP phone terminal 1-2B performs INVITE reception (S103). TheIP phone terminal 1-2B transmits 200 OK to the SIP server 2 (S104). TheSIP server 2 transmits 200 OK to the IP phone terminal 1-1A (S106). TheIP phone terminal 1-1A transmits ACK to the SIP server 2 (S107). The SIPserver 2 transmits ACK to the IP phone terminal 1-2B (S108). The IPphone terminal 1-2B performs ACK reception (S109).

The IP phone terminal 1-2B performs random number generation processing(S111). The IP phone terminal 1-2B transmits a notification of a randomnumber to the IP phone terminal 1-1A (S112). The IP phone terminal 1-2Btransitions to wait for a response (S113). The IP phone terminal 1-1Areceives key inputs, and transmits a response to the IP phone terminal1-2B (S114). The IP phone terminal 1-2B performs determination (S116).Since the determination is OK here, the IP phone terminal 1-2B plays aring tone (S117). The IP phone terminal 1-2B receives off-hook (S118).Here, the IP phone terminal 1-1A and the IP phone terminal 1-2Btransition to a talk state.

The IP phone terminal 1-2B detects an on-hook (S119). The IP phoneterminal 1-2B transmits BYE to the SIP server (S121). The SIP server 2transmits BYE to the IP phone terminal 1-1A (S122). The IP phoneterminal 1-1A transmits 200 OK to the SIP server 2 (S123). The SIPserver 2 transmits 200 OK to the IP phone terminal 1-2B (S124).

In the INVITE reception processing (S103), when an INVITE message isreceived, the incoming time 181 is set to an INVITE message receptiontime “2009/1/10 11:34:5,” the status 183 is set to “wait for ACK,” thecaller information is set to a telephone number “045-111-1111” and anSIP URI “user-1@A,” and “IP4=10.0.0.1,” and the status 183 is set to“wait for ACK” on Record 4 (#4) of the session management table 18 ofFIG. 4A.

In the ACK reception processing (S109), the status 183 of #4 of thesession management table 18 of FIG. 4B is updated to “random numbergeneration.” In the random number generation processing (S111), a timerwhich is set up arbitrarily is started, the number of digits of a randomnumber to be transmitted is determined, and the random number isdetermined. In the random number generation processing, the status 183of the session management table 18 of FIG. 4C is updated to “wait for aresponse,” and a generated random number is stored in the random number186. A random number is used in determination processing.

In the wait for a response (S113), wait for a response continues untilexpiration of the activated timer. In the determination processing, thedetermination of whether a PB sound of a received response matches avalue described on the random number 186 is performed. Upon receiving aresponse, the IP phone terminal 1-2B updates the status 183 of FIG. 4Dto “determination in process.” Upon starting to transmit a ring tone, asshown in FIG. 4E, the IP phone terminal 1-2B updates the status 183 to“ringing.” The IP phone terminal 1-2B updates the status 183 to “phonecall,” as shown in FIG. 4F. Upon detecting on-hook, the IP phoneterminal 1-2B inputs “2009/1/10 11:44:5” in the disconnection time 182as shown in FIG. 4G, and changes the status 183 to “disconnection.”

The INVITE message of the SIP is explained with reference to FIGS. 7Aand 7B. In FIGS. 7A and 7B, the INVITE message includes a request line,a Via header, a Max-Forwards header, a From header, a To header, aCall-ID header, a CSeq header, a Contact header, a Contact-type header,a Contact-Length header, and a session description using SDP. Thesession description includes a v-line, an o-line, an s-line, a c-line, at-line, an m-line, and an a-line.

INVITE is described in the request line. The Via header describesinformation about a channel through which a request passes. TheMax-Forwards header describes the maximum number of transfers and isdecremented for each transfer. The From header describes a telephonenumber and SIP URL of a caller. The To header describes a telephonenumber and SIP URL of a callee. The Call-ID header describes a callernumber. The CSeq header describes a sequence number and a method (INVITEin this case). The Contact header indicates a URL for direct contact.The Contact-type header describes SDP.

The v line describes a version 0 of the SDP. The o line describes asession owner and a session ID. The s line describes a session name. Thec line describes connection information. The t line describes a timewhen a session activates. The m line is a media line and describes asound of PCM. The a line is an attribute line.

An automatic voice response table 19 is explained with reference to FIG.8. In FIG. 8, the automatic voice response table 19 includes, as voicesounds, male child 191, female child 192, male adult 193, female adult194, and background sound 195. The voice sound is played to read aloud arandom number. The voice sound is changed for each digit of a randomnumber. The background sound is superimposed on the voice sounds. Thebackground sound is changed for each notification. To prevent automaticvoice recognition at an caller IP phone terminal, the voice sound ischanged for each digit of a random number and the background sound issuperimposed on the voice sounds.

With reference to FIGS. 9 to 12, processing of a callee IP phoneterminal is explained in detail. First, the INVITE processing isexplained with reference to FIG. 9. In FIG. 9, the IP phone terminal 1which has received an INVITE message extracts caller information from aFrom line of the INVITE message, and sets the caller information 184 ofthe session management table 18 (S11). The IP phone terminal 1 sets thestatus 183 to “wait for ACK” (S12). The IP phone terminal 1 transmits anOK message to the caller IP phone terminal (S13), and ends theprocessing.

The ACK reception processing is explained with reference to FIG. 10. TheIP phone terminal 1 which has received ACK responsive to the OK messageupdates the status 183 of the session management table 18 to “randomnumber generation” (S21), and ends the ACK reception processing.

The random number generation processing is explained with reference toFIGS. 11A to 11F. In FIGS. 11A to 11F, the IP phone terminal 1 which hasended the ACK reception processing starts a timer (S31). The IP phoneterminal 1 determines the number of digits of a random number (FIG. 11B,S32). The IP phone terminal 1 determines a random number (FIG. 11C,S33). The IP phone terminal 1 determines sampling sound voices (FIG.11D, S34). The IP phone terminal 1 selects a background sound (FIG. 11E,S36). The IP phone terminal 1 superimposes the selected background soundand the random number using the selected sampling sound voices, andtransmits a notification of the automatic voice response to an caller IPphone terminal (FIG. 11F: S37). The IP phone terminal 1 updates thestatus 183 of the session management table 18 to “wait for a response,”registers the determined random number in the random number 186 (S38),and ends the processing.

The wait for a response is explained with reference to FIG. 12. In FIG.12, the IP phone terminal 1 which has transmitted the notification ofthe random number checks the status 183 of the session management table18 (S41). The IP phone terminal 1 determines whether the status 183shows “wait for a response” (S42). The IP phone terminal 1 determineswhether the timer is in within a specified time (S43). When thedetermination is YES, the IP phone terminal 1 determines whether theresponse has been received (S44). When the determination is YES, the IPphone terminal 1 updates the status 183 to “determination in process”(S46), and ends the processing.

When the determination is NO at Step 42 (S42), the IP phone terminal 1records a status error on the comment 185 of the session managementtable 18 (S47), and ends the processing. When the determination is NO atStep 43, the IP phone terminal 1 records a timeout error on the comment185 of the session management table 18 (S48), and ends the processing.When the determination is NO at Step 44, the wait processing returns toStep 41, and continues.

The determination processing is explained with reference to FIG. 13. InFIG. 13, the IP phone terminal 1 which has received the responsedetermines whether the comment 185 of the session management table 18shows an error (S50). When the determination is YES, the IP phoneterminal 1 transitions to Step 59 (S59) mentioned later. When thedetermination is NO at Step 50, the IP phone terminal 1 compares thereceived numerical value with the random number (S51). The IP phoneterminal 1 determines whether the value and the random number match oneanother (S52). When the determination is YES, the IP phone terminal 1sets the status 183 to “ringing,” plays a ring tone (S53), and ends theprocessing. When the determination is NO at Step 52, the IP phoneterminal 1 counts invalid inputs (S54). The IP phone terminal 1determines whether a frequency of the invalid inputs is within apredetermined frequency (S55). When the determination is YES, the IPphone terminal 1 transmits a message to prompt a valid input (S56). TheIP phone terminal 1 sets the status 183 to “wait for a response” (S57),and ends the processing. When the determination is NO at Step 55, the IPphone terminal 1 registers “no valid input” in the comment 185 of thesession management table 18 (S58). The IP phone terminal 1 sets thestatus 183 to “disconnection” (S59). The IP phone terminal 1 transmits“480 Temporarily Unavailable” to the caller terminal (S61), and ends theprocessing.

Finally, with reference to FIG. 14, a processing sequence when an callerIP phone terminal transmits SPIT is explained. Here, the same IP addressand SIP URI as in FIG. 6 are used. In FIG. 14, Step 201 to Step 212 arethe same as Step 101 to Step 112 of FIG. 6, and are not explained.

On the other hand, it is difficult for an IP phone terminal 1-1B torespond to a notification of a random number. Therefore, the IP phoneterminal 1-2B terminates at the wait for a response (S213) due to atimeout error. The IP phone terminal 1-2B determines NG in thedetermination processing (S216). The IP phone terminal 1-2B transmits“480 Temporarily Unavailable” to the SIP server 2 (S217). The SIP server2 transmits ACK to the IP phone terminal 1-2B (S218). The SIP server 2transmits “480 Temporarily Unavailable” to the IP phone terminal 1-1B(S219). The IP phone terminal 1-1B transmits ACK to the SIP server 2(S220), and ends the processing.

Embodiment 2

In Embodiment 2, an automatic voice response is not performed inresponse to incoming calls registered in a previously created whitelist.Communications are blocked in response to incoming calls registered in apreviously created blacklist.

With reference to FIG. 15, the main portion of an IP phone terminal 1Ais explained. In FIG. 15, the IP phone terminal 1A includes the lineinterface (IF) 11, the processor 12, the call control portion 13, theprotocol processing portion 14, the session management table 18, theautomatic voice response table 19, a registered-user management table30, a filtering setting table 31, as well as the user interfacesincluding the speaker 15, the microphone 16, the keys 17, and the voiceprocessing circuit 10. The call control portion 13 includes theregistration portion 131, the incoming-call authorization determinationportion 133, the connection-disconnection portion 132, and the randomnumber generation portion 134.

The IP phone terminal 1A includes the registered-user management table30 and the filtering setting table 31 in addition to the components ofthe IP phone terminal 1 of FIG. 2.

With reference to FIG. 16A, a filtering setting #1 of the filteringsetting table 31 is explained. The filtering setting #1 is specified bya user of the IP phone terminal 1. In FIG. 16A, the filtering setting #1includes a target INVITE message 311, a selection mark 312, and aselection item 313. A callee sets an action in advance when there is anincoming call registered in the IP phone terminal 1. Specifically, whenan INVITE message is received from an caller through other than an SIPserver specified by the IP phone terminal 1, the callee selects one ofauthorization and rejection of the incoming call of the selection items313. In this example, the selection mark 312 is marked on the rejection.Similarly, in a filtering setting #2 of a filtering setting table 32 ofFIG. 16B, when there is an incoming call whose number is not identified,the callee selects one of authorization to, rejection of, and automaticresponse to the incoming call on the selection item 323. Here, aselection mark 322 has been marked on a random number generationresponse. Also in a filtering setting #3 of a filtering setting table 33of FIG. 16C, when there is an incoming number-identified phone call, thecallee selects one of authorization to, rejection of, and automaticresponse to the incoming call of selection items 333. Here, a selectionmark 332 has been marked on the authorization. The determination ofwhether a call has been received through an SIP server is latermentioned with reference to FIG. 18. The presence of the numberidentification is checked with reference to the From header of FIG. 7B.In FIG. 7B, the From header shows “anonymous,” not identifying a callernumber.

With reference to FIG. 17, the registered-user management table 30managing the registration of the whitelist for call authorization andthe blacklist for call rejection that are set in the phone terminal 1 isexplained. In FIG. 17, the registered-user management table 30 includesa record number 200, a caller's telephone number 201, an SIP URI 202, anIP address 203, and a list type 204.

The IP phone terminal 1 transmits 200 OK, receives ACK, and plays a ringtone immediately in response to an INVITE message including thetelephone number 201, SIP URI 202, and IP address 203 that all match thedescription of the whitelist.

On the other hand, the IP phone terminal 1 transmits “480 TemporarilyUnavailable” immediately in response to an INVITE message including thetelephone number 201, SIP URI 202, and IP address 203 that all match thedescription of the blacklist.

An SIP frame format is explained with reference to FIGS. 18A and 18B. InFIGS. 18A and 18B, an SIP frame format 70 includes a source address 71,a destination address 72, a protocol and port number 73, and an SIPINVITE message 74. In FIG. 18A, the source address 71 matches an address20.0.0.240 of an SIP server. In FIG. 18B, the source address 71 is10.0.0.3/24, and differs from the address 20.0.0.240 of the SIP server.

With reference to FIG. 19, the sequence among a caller IP phone terminalregistered in the blacklist, an callee IP phone terminal, and an SIPserver is explained. In FIG. 19, the IP phone terminal 1-1 n transmitsan INVITE message to the SIP server 2 (S301). The SIP server 2 transmitsthe INVITE message to the IP phone terminal 1-2B (S302). The IP phoneterminal 1-2B performs server determination processing (S303). The IPphone terminal 1-2B performs INVITE reception processing (S304). Here,since the IP phone terminal 1-1 n is registered in the blacklist in theINVITE reception processing, the IP phone terminal 1-2B transmits “480Temporarily Unavailable” to the SIP server 2 (S306). The SIP server 2transmits ACK to the IP phone terminal 1-2B (S307). The SIP server 2transmits “480 Temporarily Unavailable” to the IP phone terminal 1-1 n(S308). The IP phone terminal 1-1 n transmits ACK to the SIP server 2(S309).

With reference to FIG. 20, the sequence among a caller IP phone terminalregistered in the whitelist, a callee IP phone terminal, and an SIPserver is explained. In FIG. 20, an IP phone terminal 1-2 m transmits anINVITE message to the SIP server 2 (S401). The SIP server 2 transmitsthe INVITE message to the IP phone terminal 1-2B (S402). The IP phoneterminal 1-2B performs server determination processing (S403). The IPphone terminal 1-2B performs INVITE reception processing (S404). Here,since the IP phone terminal 1-2 m has been registered in the whitelistin the INVITE reception processing, the IP phone terminal 1-2B transmits200 OK to the SIP server 2 (S406). The SIP server 2 transmits 200 OK tothe IP phone terminal 1-2 m (S407). The IP phone terminal 1-2 mtransmits ACK to the SIP server 2 (S408). The SIP server 2 transmits ACKto the IP phone terminal 1-2B (S409). The IP phone terminal 1-2Bperforms ACK reception processing (S411). The IP phone terminal 1-2Bplays a ring tone (S412). The IP phone terminal 1-2B detects off-hook(S413), and starts communication.

With reference to FIG. 21, the sequence among a caller IP phone terminalnot registered in a whitelist/blacklist, a callee IP phone terminal, andan SIP server is explained. In FIG. 21, the IP phone terminal 1-1Atransmits an INVITE message to the SIP server 2 (S501). The SIP server 2transmits the INVITE message to the IP phone terminal 1-2B (S502). TheIP phone terminal 1-2B performs server determination processing (S503).The IP phone terminal 1-2B performs INVITE reception processing (S504).Here, since the filtering setting of FIG. 16 is set such that the randomnumber generation response is selected when a call whose number is notidentified is detected in the INVITE reception processing, the IP phoneterminal 1-2B transmits 200 OK to the SIP server 2 (S506). The SIPserver 2 transmits 200 OK to the IP phone terminal 1-1A (S507). The IPphone terminal 1-1A transmits ACK to the SIP server (S508). The SIPserver 2 transmits ACK to the IP phone terminal 1-2B (S509). The IPphone terminal 1-2B performs ACK reception processing (S511). The IPphone terminal 1-2B starts a timer, and performs random numbergeneration processing (S512). The IP phone terminal 1-2B transmits anotification of a random number to the IP phone terminal 1-1A (S513).The IP phone terminal 1-2B starts wait for a response (S514). The IPphone terminal 1-1A receives key inputs (S516). The IP phone terminal1-1A transmits a response of a PB signal changed from the random numberto the IP phone terminal 1-2B (S517). The IP phone terminal 1-2Bperforms determination processing of the PB signal and the transmittedrandom number (S518). Here, since both match, the IP phone terminal 1-2Bplays a ring tone (S519). The IP phone terminal 1-2B detects off-hook(S521), and starts communication.

The server determination processing is explained with reference to FIG.22. In FIG. 22, the IP phone terminal 1 extracts a caller IP address(S71). The IP phone terminal 1 determines whether the reception is froman SIP server (S72). The IP phone terminal 1 checks the filteringsetting when the determination is NO (S73). The IP phone terminal 1determines whether to reject the caller (S74). The IP phone terminal 1registers “disconnection” in the status 183 of the session managementtable 18 when the determination is YES (S76). The IP phone terminal 1records “reception from other than a specified server,” and the callerIP address on the comment 185 of the session management table 18 (S77).The IP phone terminal 1 transmits “480 Temporarily Unavailable” to thecaller IP phone terminal (S78), and ends the processing. The IP phoneterminal 1 ends the server determination processing when thedetermination is YES at Step 72 and NO at Step 74.

The INVITE reception processing is explained with reference to FIG. 23.In FIG. 23, the IP phone terminal 1 extracts caller information from theFrom header, and registers the caller information in the sessionmanagement table 18 (S81). The IP phone terminal 1 determines whether acaller phone number is notified (S82). The IP phone terminal 1 checksfiltering setting #2 when the caller phone number is not notified (S84).The IP phone terminal 1 determines whether to reject the caller (S86).The IP phone terminal 1 updates the status to “disconnection” when thedetermination is YES (S87). The IP phone terminal 1 transmits “480Temporarily Unavailable” (S88), and ends the processing.

The IP phone terminal 1 searches for the registered-user managementtable 30 when the caller phone number is notified at Step 82 (S83). TheIP phone terminal 1 determines whether the list type for the number isthe blacklist (S85). When the determination is YES, the IP phoneterminal 1 registers in the remark “reception from the caller in theblacklist” (S89), and transitions to Step 87.

When the determination is NO at Step 85, the IP phone terminal 1determines whether the list type for the number is the whitelist (S90).When the determination is YES, the IP phone terminal 1 registers in theremark field “reception from the caller in the whitelist” (S91). The IPphone terminal 1 sets the status to “wait for ACK” (S92). The IP phoneterminal 1 transmits OK (S93), and ends the processing.

The IP phone terminal 1 checks the filtering #3 when the determinationis NO at Step 90 (S94). The IP phone terminal 1 determines whether toreject the caller (S86). The IP phone terminal 1 transitions to Step 87when the determination is YES. The IP phone terminal 1 determineswhether to authorize the caller at Step 86 when the determination is NO(S96). When the determination is NO, the IP phone terminal 1 changes thestatus to “random number generation” (S97), and transitions to Step 93.The IP phone terminal 1 transitions to Step 92 when the determination isYES at Step 96.

The ACK reception processing is explained with reference to FIG. 24. InFIG. 24, the IP phone terminal 1 checks the status (S66). The IP phoneterminal 1 determines whether the status shows wait for ACK (S67). Whenthe determination is YES, the IP phone terminal 1 plays a ring tone(S68), and ends the processing. The IP phone terminal 1 ends theprocessing without change when the determination is NO at Step 67.

With reference to FIGS. 25A to 25C, transition of the session managementtable when setting the filtering is explained. In FIG. 25A, Record of #1shows a result of the server determination processing after receiving anINVITE message from other than an SIP server. Because the INVITE messagehas been determined to be received from other than a server registeredin the IP phone terminal 1 in the server determination processing,“disconnection” has been registered in the status, and “incoming callfrom other than an SIP server” and the caller IP address have beenregistered in the comment field. Record of #2 shows a result of theINVITE reception processing after receiving an INVITE message from acaller in the blacklist. An incoming time and caller information havebeen registered in the INVITE reception processing. Further, since thecaller has been registered in the blacklist, “disconnection” has beenregistered in the status and “incoming call from the caller in theblacklist” has been registered in the comment field. Record #3 shows aresult of the following processing: reception of the INVITE message froma caller in the whitelist; registration of an incoming time and callerinformation in the INVITE reception processing; registration of“incoming call from a caller in the whitelist” in the comment fieldbecause the caller has been registered in the whitelist; communicationafter registering wait for ACK in the status; and registration of adisconnection time and registration of “disconnection” in the statusafter the communication. Record #4 shows a result of the INVITEreception processing after receiving the INVITE message from other thancallers of the white/blacklist. An incoming time and caller informationhave been registered at the time of INVITE reception processing, andfurther, since a number of this incoming call has not been identified,“random number generation” has been registered in the status based onthe filtering setting #2. As shown in FIG. 25B, on Record #4, the statusis changed to “wait for a response” and a random number is registeredafter the ACK reception and random number generation processing. Then,the status is changed to “ringing” after wait for a response anddetermination processing, and a ring tone is played. As shown in FIG.25C, when off-hook is detected, the status is set to “phone call.”

What is claimed is:
 1. A selective response unit used for IP telephoneservices comprising: a call control portion; a line interface portion; amemory portion recording thereon an automatic voice response table; anda voice processing circuit; wherein the call control portion includes anincoming-call authorization determination portion, aconnection-disconnection determination portion, and a random numbergeneration portion; the incoming-call authorization determinationportion, when receiving a call connection request, is configured toobtain a first voice corresponding to a random number generated in therandom number generation portion from the automatic voice responsetable, and to transmit the first voice from the voice processing circuitto a source of the call connection request; the connection-disconnectiondetermination portion, when the voice processing circuit does notreceive a number equal to the random number from the source within apredetermined time, is configured to transmit a call disconnectionmessage to the source; the memory portion is further configured torecord thereon a registered-user management table and a filteringsetting table; and the incoming-call authorization determinationportion, when receiving a call connection request, is configured toselect any one of transmission of the first voice, transfer of a callconnection request or transmission of an incoming call signal, andtransmission of a call disconnection message based on theregistered-user management table and the filtering setting table.